The quality has to do with the number of increments avalible and the sampling rates nyquist cut off.

Higher sampling rates restore some of the ultra high frequency harmonics that are higher in pitch than we can hear that gives each instrument more of its "distinctive" quality.

If you are doing CD format (know as redbook) finals, then they are at 16 bit 44.1K and their is therory that, (in my experience) recording at higher frequencies will change the shapes of the waves we do hear and trickle down to the redbook or CD standard.

The higher the frequency, the more "snippits per second" of information is stored and the higher fidelity assumed. This is most apparent when using multiple tracks in digital.

Face it. What sounds better to you?

I have had engineers swear by keeping everything at 44.1 since it will end up that way, assuming that their sample rate converters cause audible problems.

Do yourself a huge favor.

Do works in all of the sampling rates you have, and burn them to cd. Pick the one you feel is closest to the actual performance.

Yes, Dusty, yes there is. It is a difference in ultra high frequency response more than anything else. While the question if this makes a difference is debated quite hotly, I had the opportunity to do an A /B test of this once and I was convinced that 96 K sounded much better than 48. The difference in openess and stereo depth was very evident. Even when you dither down to 44.1 for CD's this can be significant through the production process as the existence of these high harmonics can excite lower more audible frequencies. I feel it is a significant improvement. If you can afford the file size and the throughput on your machine it is a worthwhile pursuit. Fats
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Tannoy, Dynaudio, Blue Sky, JBL, Earthworks, Westlake, NS 10's , Genelec, Hafler, KRK, and PMC
Those are good. …………………….. Pick one.
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John,
My Internet computer and my ISP are woefully inadequate for this application. Secondly this test was performed a couple years ago with client material and I don't have the tapes, client took them. Third, IMO any Internet transfer sucks so bad I doubt that the quality would show. IMO it's something you have to do in a studio. This test was performed at my old studio KFRS in Fremont CA using an analog 2" multitrack through an Apogee 24 /96 PSX 100. (See old thread in Small Steps “Why Digital Still Sucks) I a/b'd the analog and then the different rates starting at 96 and coming down to 48 the 44.1. Each step showed a further loss in quality. Bill may be able to post something like this. He has a much better Internet interface than I, however I am still skeptical if the quality differences world be evident on mp3’s. ... Fats
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Tannoy, Dynaudio, Blue Sky, JBL, Earthworks, Westlake, NS 10's , Genelec, Hafler, KRK, and PMC
Those are good. …………………….. Pick one.
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Fats, you cannot stream with a modem but you can download the file and I have mp3's that are very close to wavs and some that were done from 24 bit that are beyond redbook. You have to have the file to play back as streaming with a modem is no good.

> Why don't you post your files as a blind A/B and we'll have a poll <

I agree with that. But the big problem with most "tests" I've seen is they compare apples and oranges. The only way to truly compare, say, 44.1 and 96 KHz. is to route one preamp output to two separate tracks - or, more likely, separate DAWs - and record the same performance at the same time. Anything less is just guessing.

I've read that, beyond the NyQuist stuff, a clear advantage of higher sampling rates is more accurate stereo localization by a listener.

From the ArtistPro website:

It’s been determined that time delay differences of 15 microseconds between left and right ears are easily discernible by nearly anyone. That’s less than the time difference between two samples at 48kHz (about 20 microseconds). Using a single pulse, one microsecond in length as a source, some listeners can perceive time delay differences of as little as five microseconds between left and right. It is therefore, indicated that, in order to provide a system with exact accuracy concerning imaging and positioning, the individual samples should be less than five microseconds apart. At 96kHz (a popularly preferred sample rate) there is a 10.417-microsecond space between samples. At 192kHz sample rate there is a 5.208-microsecond space between samples. This reasoning suggests that a sample rate of 192kHz is probably a good choice.

Click to expand...

I'm interested, as the first poster was, if anyone can hear the difference between 96 and 192. That is, should someone looking to buy a 24-bit system today bite the bullet and get 192, or will 96 become the standard for a few years?

The advantages of the other leaps in sample rates seem well established.

there's a difference of course...but not enough to worry about in that things will always continue to "progress" and the interelationship of all of the variables that go intop making great recorded music make the format choice just that...a choice...not the make or break issue.
"Jagged little Pill" weather you liker it or not was a sucessful record on many levels and recorded and mixed off of blackface adats.
"Sgt. Pepper's..." was only(!). four track analog.
So don't wait for the next big thing....do something now with the best that you have at hand...the best way YOU can...becuase you more than the medium dictate the results.

Originally posted by RecorderMan: So don't wait for the next big thing....do something now with the best that you have at hand...the best way YOU can...becuase you more than the medium dictate the results.

Click to expand...

Very well said RecorderMan. Even bleeding edge technology can't compensate for one that doesn't know how to get good results; or know what good results are for that matter

When an analog to digital converter (ADC) is operating at 44,100 samples per second, it can't 'see' any wave form that is higher than 22,050hz (nyqyist). If it does encounter a situation where there is a frequency greater than 22,050hz,
aliasing occurs which causes distortion across the frequency band. To prevent this from happening, a low-pass filter is installed in every ADC. The actual reason higher sampling rates sound better is because they can accurately convert the signals that would otherwise cause distortion.

What this means is that the higher the sampling rate of the ADCs the better, but once it's been converted the only thing that affects the quality is the dithering process (which can be virtually transparent depending on the algorithm).

The other thing to note is the bit depth. A 16 bit converter can only distinguish 65,536 different levels per sample, where a 20 bit one has 1,048,576 leves, and 24 bit has 16,777,216 different levels. Although you may be able to distinguish a small bit of difference between 16 bit and 20 bit, there is probably no noticable difference between 20 bit and 24 bit.

The caveat to all of this is that different converters have different quality, so one brand of 44khz/16bit converters may sound better than another brand of 192khz/24bits. It also depends on the clock source for the converters, but that's for a whole other thread. I hope I didn't confuse anybody too badly.

The biggest difference that is audible is the resolution of the recording. Going from 16 bit to 24 bit will show a higher degree of fidelity than the sampling rate.

A 48 KHz sampling rate translates into a response up to 24 KHz & A 96 KHz sampling rate translates into a response of 0-48 KHz, well beyond our hearing range. 192 KHz translates into a response curve up to to 96 KHz. I would go with the 96 KHz sampling rate captures more harmonics and is much easier on our hearing. It really depends on what you are recording also. Cymbals for example have a natural decay all the way up to 50 KHz. The digital realm employs sharp filters. For example a sampling rate of 48KHz has a smooth response to 22 KHz. There is a very steep curve from 22 KHz to 24KHzI believe analog does a much better job of capturing the recording and then dumping it or slaving it into the digital realm, especially with older digital equipment. A/D converters have improved exponentially over the years. If you have the space, record at the highest resolution & then dither it down to the required format such as 44.1 KHz, 16 bit.

One thing that never seems to get addressed when we talk/bitch about converters and sampling rates is the converter's front end and it's componentry. If you build an A/D converter out of 10% tolerance parts on the input side, it's not going to sound as good as something built with 1% tolerance components. Of course, those parts cost a lot more.

We all get excited when we hear class A circuits. No kidding. They're built to a high spec! Many of the analog pieces we lust after were carefully designed and tweaked so that they were either sonically pleasant or as transparent as possible with the technology of the time. Both seems to happen a lot with tube gear and analog tape.

I wager, though, that if you recorded with good pre's through a high quality D/A (i.e. Apogee, Prism, etc) clocked with a high quality master clock (Nanosyncs, Aardsynch, etc) you'd be very impressed with the warm and punchy sound of your rig.

I love tape. I have tape decks here and always try to figure out how to use them on things. I love the analog process moreso than holding on to it as the holy grail of sound. Sorry guys, analog has it's limitations. It's expensive, more difficult to edit, requires lots of maintenance, has a higher noise floor and less dynamic range, and is generally more physically delicate than a hard-disk system in some ways. But GOD does it sound good! Heh!

Digital is more of a wysiwyg (should be wyHiwyg, h=hear). What you put into it is largely what you get out of it. The systems were designed to be linear and have flat response across the audible spectrum. Analog certainly doesn't do that. That's why we love it so much. Same as an overdriven tube. Sounds yummy, but certainly isn't linear. I'm not arguing for one or the other here, just trying to present the strenghts and weaknesses.

Seems to me the time we spend tracking to analog tape and then transferring to digital would be better spent figuring out how to get the results we want from our digital rigs. Oh, and for those of you that cite "Sgt Peppers was just 4-track". Yeah. They used 2 4 tracks which was high tech back then. These were not portastudios, guys. They were some of the best machines in the world, with some of the best consoles/other gear in the world. It's not about the track count on that one. 1" 4 track has some FAT track width, track width=resolution.

A couiple of observations.
1. If someone want to a/b an audio file you had best use something that has not experienced data compression (like an mp3) or you are going to be testing the mp3 encoder more than the source material.

2. As Dan said, the quality of the analog front end to the converter is the single most important sonic element in an A/D converter. The next most important thing is the stability of the master clock. A very stable clock will provide a much better sound stage.

3. So far there has been no documented tests or proofs that higher sample rate makes any difference in the audible quality of a digital recording. For one thing it is almost impossible 9to make a fair test) given that a 96ksps system will have a completely different set of analog filters in fron of it than a 48ksps system will. I have a 192i/o and great monitors. I cannot tell the difference so far.

I am not saying there is no difference but so far no one has been able to publish anything other than speculation and physics does not support there being a difference.

Again i am only referring to sample rate influence. It is also possible by the way that higher sample rate designs can degrade performance. This occurrs when the designer decides to use a cheaper or sloppier anti
aliasing filter design for the 96ksps system.

I have personally done extensive analysis in the DSP realm of digital audio and I can tell you that an impule response sampled at 48ksps and 96ksps will have the EXACT same spectrum within the Nyquist pass band..

I believe the AES has acommittee looking into designing something that might be a fair testing process. Until then, the only reason I can see to buy higher sample rate converters is :
1. the cost the same as lower rate converters
2. if you need to provide content for DVD-A
3. you just gotta have the biggest,fastest widget on the block.

One thing to keep in mind is, if there's any chance you'll mix digitally, you should keep your sample rate at a multiple of 44.1k. IE 44.1, 88.2, or 176.4. That way when downsampling to 44.1 for CD no fancy sample rate conversion math is reqired, it's just a matter of throwing away every other sample. Sample rate conversion of non-integer ratios just never sounds good.

PS somebody earlier said something about "dithering" down to 44.1; just to keep our terminology straight, dither has nothing to do with sample rates- it's used sometimes in reducing BIT DEPTH.

All this theory makes sense. I think a question for a lot of us "semi-pros" here is will our lower end daws have an audible difference withthe higher sampling rates? (In my case a pair of delta 1010LT's for 16 tracks. Up to at least 96khz is available)

My second question is the clock. If we got a high quality clock (Say apogeee etc...) would it make an audible improvement as well? Or since the front end is so cheap to beign with it wouldn't really matter and money would be better spent elsewhere?

Originally posted by Paul Berolzheimer: PS somebody earlier said something about "dithering" down to 44.1; just to keep our terminology straight, dither has nothing to do with sample rates- it's used sometimes in reducing BIT DEPTH.

Click to expand...

Dithering doesn't just apply to bit depth. It's a generalized term meaning that a reduction in data (quality) is required. This happens when converting to a lower bit depth and/or sampling rate.